Cisco 9971 phone configuration working example with setup tips

Sorry for that, may be you have some experience about why conference button not working? every thing is fine calls, transfer, etc. problems only with conference

1 Like

Hi

Thanks for your work and support.

Iā€™m trying to get my cisco 9971 to register with asterisk (11.3.0)+freePBX (2.11.0.0beta2.8) running on Raspberry Pi (http://www.raspberry-asterisk.org/)

my question here is: How to patch my (installed and running) asterisk on R.Pi with the files on:

https://issues.asterisk.org/jira/browse/13996

Iā€™ve figured (yet Iā€™m not sure) that I only need to apply the ā€œgareth-11.3.0.patchā€ patch. I donā€™t know if that is all I need.

There are also two patches called: video2-new.patch and smime.p7s on this website linked at:

http://lists.digium.com/pipermail/asterisk-video/2011-October/003560.html

but Iā€™m not sure if they are already included in the latest gareth-11.3.0.patch or how to know for sure if they are or are not.

If they are not then Iā€™ll need to also apply them. I donā€™t know if should I apply them before or after gareth-11.3.0.patch

but it seems to me that those 2 patches (video2-new.patch and smime.p7s) are for asterisk-1.8.4.4!! I hope gareth-11.3.0.patch is enough.

Iā€™m trying to follow your explanation on how to configure 9971 cisco phone to register with asterisk but till now was not successful. I have yet to see a registration attempt from asterisk Cli.

on the phone web log I found some where that it tried to register sip (correct sip to correct asterisk ip) but did not get a respond.

my cisco 9971 FW is sip9971.9-2-3-27. I have the update to 9.3.1.33 but Iā€™ve figured it is better to wait to make sure before updating just in case the older firmware was more reliable/asterisk friendly as i donā€™t know how to downgrade the FW after updating.

I think I have got the phone configuration file right and only added 2 sip extensions to be utilized in the phone and configured it in the SEP(mac_address).cnf.xml file but still phone is not registering. so I guess there is no more path for me to go except patching asterisk and to hope that this is all I need to get it to work.

another question about the NTP

is it:
D/M/Ya
or
D/M/YA
or
D/M/Y

as I still didnā€™t get my clock to update with NTP: 203.88.112.222

more details about my settings/configurations will be further gladly provided if needed.

thanks for your help.

Did you ever solve this,
I note my phones seem to be ignoring the
video_imageattr=recv [x=640,y=480,q=0.50]
and using 176x144

Iā€™m willing to even pay someone for some support to get my PBX up and running, I think I have the configuration file built correctly but my phone keeps saying it canā€™t download configuration file from 192.xxx.x.xx . Iā€™m still a linux noob in a way so any help would be appreciated and if I need to pay for support to learn this I will, this is more or less just going to be for personal gain / use, Iā€™m a nerd I love to play with technology but there are times it gets a upper hand on me :stuck_out_tongue:

So got my phone today and trying for the life of me to get it work but I get the usual Error Verifying Config Info, Iā€™ve pretty much taken the sample in this post and tried even just doing a copy and paste with getting this same error. I then tried to remove some extra items that I wouldnā€™t use like speed buttons etc and same issue,

http://pastebin.com/jfqZqbjp

I cant for the life of me get the code to function correctly on here so Iā€™ve put it in a pastebin

Well scrap my previous question got the phone to register, so the only two features I canā€™t figure out still are how to get to missed calls / contact book. That pastbin config is no longer valid I modified it a few times and got it to work.

So question 1) When I hit the address book button next to the settings gear it says service not available, so how would I get it to show local services (Address Book, Missed Calls, Placed Calls)

and 2) does conference calling on the phone work yet?

Please post your new working config itā€™s something very popular.

I donā€™t think the conferencing has been fixed, I donā€™t know I still canā€™t get mine to register so I am looking forward to your post.

The Cisco directory is a web page, do you know how to add content to httpd (apache) web server? If so Google Cisco Phone Apps, there are many free scripts out on the net.

I can get my phone to register now as well, placed a call with my google line and was able to call out, having problems getting my incoming calls on google to work though, I had it working once but have been unable to get it to work since I reinstalled. I tried using the forwarding script in here to but it seems theres a bug with it, but so far Iā€™m figuring out most of the stuff, contacts Iā€™m still working on as well, and a few others, Iā€™m guessing there in the features.conf but I have yet to locate an example one that I can go off of.

So are either of you guys going to share a working config?

Many thanks for posting this tutorial. I got my Cisco 8961 working properly.
The only part I still have issues is WMI. I use voip.ms as provider and for some reason there is no notification on my device for new messages.
[general]
ā€¦
mwi => 123456:[email protected]:5060/101

Just to note I couldnā€™t get my phone to download from the tftpboot folder on my freepbx server, I ended up downloading a free open source tftp server for my windows 7 box and transferring firmware/config files to my phone that way.

Here is my main config file labled SEPMACADDRESSOFPHONE.cnf.xml Note this has to be all caps or the phone wonā€™t take it.

<?xml version="1.0" encoding="UTF-8"?> SIP

admin
cisco

D/M/Ya Mountain Standard/Daylight Time 198.60.73.8 Unicast
 <callManagerGroup>
    <members>
       <member priority="0">
          <callManager>
             <ports>
                <ethernetPhonePort>2000</ethernetPhonePort>
                <sipPort>5060</sipPort>
                <securedSipPort>5061</securedSipPort>
             </ports>
             <processNodeName>IP ADDRESS OF YOUR PBX</processNodeName>
          </callManager>
       </member>
    </members>
 </callManagerGroup>
true 3

SIP8941_8945.9-3-2-12
DefaultFP.xml

false false 0 1 0 0 1 1 0,1,2 1 0 0,1 0 0 0 0 1,2,3,4,5,6,7 08:30 09:30 01:00 1 1 1 1 1 1 0,1,2 1 0,1 1 0,1 0 1 1 1 0,1,2 1 1 1 0,1 0 1 0 2 0

United_States

<networkLocaleInfo> 
	<name>English_United_States</name> 
	<uid>64</uid> 
	<version>1.0.0.0-1</version> 
</networkLocaleInfo> 

1

0








96
0
96

2
5
1
0
0
1
false
0
0

0


3804


false


USECALLMANAGER
5060
USECALLMANAGER
5060


true

 <sipCallFeatures>
    <cnfJoinEnabled>true</cnfJoinEnabled>
    <callForwardURI>x-cisco-serviceuri-cfwdall</callForwardURI>
    <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
    <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
    <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
    <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
    <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
    <rfc2543Hold>false</rfc2543Hold>
    <callHoldRingback>2</callHoldRingback>
    <localCfwdEnable>true</localCfwdEnable>
    <semiAttendedTransfer>true</semiAttendedTransfer>
    <anonymousCallBlock>2</anonymousCallBlock>
    <callerIdBlocking>2</callerIdBlocking>
    <dndControl>0</dndControl>
    <remoteCcEnable>true</remoteCcEnable>
<retainForwardInformation>true</retainForwardInformation>
 </sipCallFeatures>

 <sipStack>
    <sipInviteRetx>6</sipInviteRetx>
    <sipRetx>10</sipRetx>
    <timerInviteExpires>180</timerInviteExpires>
    <timerRegisterExpires>3600</timerRegisterExpires>
    <timerRegisterDelta>5</timerRegisterDelta>
    <timerKeepAliveExpires>120</timerKeepAliveExpires>
    <timerSubscribeExpires>120</timerSubscribeExpires>
    <timerSubscribeDelta>5</timerSubscribeDelta>
    <timerT1>500</timerT1>
    <timerT2>4000</timerT2>
    <maxRedirects>70</maxRedirects>
    <remotePartyID>false</remotePartyID>
    <userInfo>None</userInfo>
 </sipStack>

 <autoAnswerTimer>0</autoAnswerTimer>
 <autoAnswerAltBehavior>false</autoAnswerAltBehavior>
 <autoAnswerOverride>true</autoAnswerOverride>
 <transferOnhookEnabled>false</transferOnhookEnabled>
 <enableVad>false</enableVad>
 <preferredCodec>none</preferredCodec>
 <dtmfAvtPayload>101</dtmfAvtPayload>
 <dtmfDbLevel>3</dtmfDbLevel>
 <dtmfOutofBand>avt</dtmfOutofBand>
 <alwaysUsePrimeLine>true</alwaysUsePrimeLine>
 <alwaysUsePrimeLineVoiceMail>true</alwaysUsePrimeLineVoiceMail>
 <kpml>3</kpml>
 <natEnabled>false</natEnabled>
 <natAddress>IP ADDRESS OF YOUR PBX</natAddress>

 <stutterMsgWaiting>2</stutterMsgWaiting>

 <callStats>false</callStats>
 <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
 <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>


 <startMediaPort>10000</startMediaPort>
 <stopMediaPort>20000</stopMediaPort>
 <voipControlPort>5060</voipControlPort>
 <dscpForAudio>184</dscpForAudio>
 <dscpVideo>136</dscpVideo>
 <dscpForTelepresence>128</dscpForTelepresence>
 <ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
 <softKeyFile>softkey.xml</softKeyFile>
 <dialTemplate>dialplan.xml</dialTemplate>
 <phoneLabel>This can be anything it seems</phoneLabel>
 <sipLines>
    <line button="1" lineIndex="1">
       <featureID>9</featureID>
       <featureLabel>Line Label on phone</featureLabel>
       <name>FreePBX Extension you setup</name>
       <displayName>Incoming Label I think?</displayName>
       <contact> FreePBX Extension you setup </contact>
       <proxy>USECALLMANAGER</proxy>
       <port>5060</port>
       <autoAnswer>
          <autoAnswerEnabled>2</autoAnswerEnabled>
       </autoAnswer>
       <callWaiting>3</callWaiting>

       <authName> FreePBX Extension you setup </authName>
       <authPassword>FreePBX Extension Secret/Password</authPassword>

       <sharedLine>false</sharedLine>
       <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
   <messageWaitingAMWI>1</messageWaitingAMWI>
       <messagesNumber>Voicemail Extension go's here</messagesNumber>
       <ringSettingIdle>4</ringSettingIdle>
       <ringSettingActive>5</ringSettingActive>

       <forwardCallInfoDisplay>
          <callerName>true</callerName>
          <callerNumber>false</callerNumber>
          <redirectedNumber>false</redirectedNumber>
          <dialedNumber>true</dialedNumber>
       </forwardCallInfoDisplay>
		<maxNumCalls>4</maxNumCalls>
		<busyTrigger>2</busyTrigger>
    </line>

    
</sipLines>
0 Missed Calls Application:Cisco/MissedCalls Voicemail Application:Cisco/Voicemail Received Calls Application:Cisco/ReceivedCalls Placed Calls Application:Cisco/PlacedCalls Personal Directory Application:Cisco/PersonalDirectory Corporate Directory (Demo) http://directory.ciscoxmlservices.com/demo1/demo1/

Next is my dialplan.xml

Heres my XMLDefault.cnf.xml

2000 5060 5061 FreePBX IP Firmware Version see below

My firmware was SIP8941_8945.9-3-2-12 now also to note
I had 10 files labeled like SIP8941_8945.9-3-2-12.bin1.sgn except bin1 would be like bin2 etc, then there was a .loads after one of them. Another thing to note is there are multiple firmware files on Ciscoā€™s site, the ones that have a SP at the end or wherever are service packs, they do not include the BOOT8941_8945.0-0-1-0.bin.sgn file which you need in order to initiate the firmware upgrade, I would recommend doing the older firmware first then the service pack. If I can help out anymore let me know, Iā€™m just slow to getting back to my posts sorry about that.

So Iā€™ve been staring at this for several hours a day over the weekend and Iā€™m missing something. I set up a new box in the lab, CentOS 6, Asterisk 11.5.1, FreePBX 2.11, chan_sccp 4 (for 7921), and loaded Garethā€™s presence patch from JIRA. No matter what I do I canā€™t get these phones to register over TCP for presence to work.

In the FreePBX GUI I have transport set to All - TCP Primary. If I set TCP only the phones wont register at all. Allowing UDP lets registrations and calls occur, but presence is ignored. Whatā€™s also strange is the configs that were shared here list UDP as the transport. This line in the SEP.XML defines UDP as the transport:
2

Iā€™ve tried using 1 and 4 here with no avail. Packet captures on the pbx show the phone doesnā€™t even make an attempt on port 5060. In UDP mode I do see the presence notification and acknowledgment. (Note, I mean SIP acknowledgement not layer 4 TCP ack flag).

22:14:06.618205 IP 10.0.10.55.sip > 10.0.5.101.sip: SIP, length: 855 E`.s....@... . 7 ..e....._'.NOTIFY sip:[email protected]:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 72.78.211.55:5060;branch=z9hG4bK746828ea Max-Forwards: 70 From: ;tag=as2ec9a175 To: ;tag=e840400d3bcc00043b34d142-171f7a7c Contact: Call-ID: [email protected] CSeq: 107 NOTIFY User-Agent: FPBX-2.11.0(11.5.1) Subscription-State: active;expires=3600 Event: presence Content-Type: application/cpim-pidf+xml Content-Length: 345 <?xml version="1.0"?>

22:14:06.641256 IP 10.0.5.101.49159 > 10.0.10.55.sip: SIP, length: 416
E`ā€¦>ā€¦P
ā€¦e
.
7ā€¦SIP/2.0 200 OK
Via: SIP/2.0/UDP 72.78.211.55:5060;branch=z9hG4bK746828ea;received=10.0.10.55
From: sip:[email protected];tag=as2ec9a175
To: sip:[email protected];tag=e840400d3bcc00043b34d142-171f7a7c
Call-ID: [email protected]
Date: Mon, 09 Sep 2013 02:14:06 GMT
CSeq: 107 NOTIFY
Server: Cisco-CP9971/9.3.4
Contact: sip:[email protected]:5060;transport=UDP
Content-Length: 0

I did set ciscounified=yes under the extensions in sip_custom_post.conf as well. Here is my XML file for the phone. Iā€™ve left everything intact as this is in a lab with no access to anything real.

SIP M/D/YA Eastern Standard/Daylight Time 0.0.0.0 unicast 5060 10.0.10.55 true true true service-uri-cfwdall service-uri-pickup service-uri-gpickup 2 true 2 2 2 true true 9 3083 USECALLMANAGER 5060 3083 2 1 3083 Passw0rd123 false 1111 5 true true true true 9 Intercom USECALLMANAGER 5060 13083 3 1 13083 Passw0rd123 false 1111 5 true true true true 21 Mr. Burns 3081 1 21 Smithers 3802 1 21 Service Queue *45 1

false
g711ulaw
dialplan.xml
softkeyDefault.xml
true
true
1
0
H. Simpson
2
true
184
136


admin
2

featurePolicyDefault.xml
sip9971.9-3-4-24.loads



1
1

admin
Lab123


2

Missed Calls

Application:Cisco/MissedCalls




Received Calls

Application:Cisco/ReceivedCalls




Placed Calls

Application:Cisco/PlacedCalls




Voicemail

Application:Cisco/Voicemail




0003298300203293

English_United_States
en

United_States

United_States



http://phone-xml.berbee.com/menu.xml
0
96
2

Hello,

Does anybody know the config file parameters needed to enable use of a KEM with the 9971 phone?
I am using a couple of these phones at home and have everything functional, with loads of help from this and other such threads I found in the community. Unfortunately, Iā€™ve not been able to discover anything about configuring or enabling a KEM.

I got a KEM for the 9971 phone and when I plug it in the phone shows a message that the LKEM has been disabled by the administrator. I found CISCO documentation describing how to enable it via Call Manager and I presume that would create some new entries in the config file for the phone. Any thoughts on how I might get a hold of those settings?

Have you just tried programming key11? I have a 9971 at home. I may just pick up the sidecar for fun.

Someone mentioned the missing dial key to me the other day. I had never seen it. Moved a 9951 from one server to the another and now the dial key is missing (actually all the feature keys) and the camera says disabled by administrator.

Very strange, no config changes made.

I tried changing the camera variables from 1 to 0 just to see a change and there was none.

If anyone know the cause of these symptoms let me know. I will keep hacking away while I watch the Cleveland Browns lose another game.

I tried programming several extra keys - 7, 8, 30, 22, etc., randomly to see if something helps. Not sure of the significance of 11 since the phone has 6 keys but I tried the 11 also now. The config file is accepted by the phone without any problems but the KEM still doesnā€™t work.

CKEM 36 Unfortunately the phone cannot power up the KEM even now - seems not to have enough power coming from a cheap AT POE injector I was using. Have ordered an "expensive" one to see if that does the trick

@simcity I got this 9971 on my desk a year ago - i followed your step that time and had it registered. Now though had to relocate and started from scratch. I am kind of spending days and night with no luck in registering the phone. Keeps saying: ā€œError Verifying Config Infoā€ although the SEPā€¦cnf.xml is a copy of yours above! Help pleaseā€¦ Ciao

@simcity I got this 9971 on my desk a year ago - i followed your step that time and had it registered. Now though had to relocate and started from scratch. I am kind of spending days and night with no luck in registering the phone. Keeps saying: ā€œError Verifying Config Infoā€ although the SEPā€¦cnf.xml is a copy of yours above! Help pleaseā€¦ Ciao

Let me express my gratitude for sharing this information.
Iā€™m currently on a quest to connect
3 cisco9951
2 cisco9971 + cams
2 2N Helios
to an AVAYA IPOFFICE 500 V2.

The information provided was paramount in resurrecting the "zombified"
cisco terminals.
Thank you !
Merry Christmas